Multiplex communications – Data flow congestion prevention or control – Control of data admission to the network
Reexamination Certificate
1999-03-17
2003-11-18
Yao, Kwang Bin (Department: 2664)
Multiplex communications
Data flow congestion prevention or control
Control of data admission to the network
Reexamination Certificate
active
06650619
ABSTRACT:
BACKGROUND OF THE INVENTION
1. Field of the Invention
The present invention relates to telecommunications transmission systems and more particularly to methods for facilitating increased call capacity in a telephony system when faced with an actual or likely surge in demand for use of the system.
2. Description of the Related Art
The invention is particularly useful in the context of internet telephony (also known as voice-over-IP (VoIP) or IP telephony), which should be understood to be a telephone system in which real-time media signals and/or data signals are communicated via a packet switched network such as the Internet, whether between two parties or between multiple parties (in a conference or multicast environment). Internet telephony may also be referred to as packet switched telephony. More generally, however, the invention may extend to use in connection with the communication of any real-time media and/or data signals over any packet switched communications link, including, for instance, IP, ATM, frame relay, X.25 and SNA networks, whether local area, metropolitan area or wide area, and point-to-point or direct end-to-end connections.
Those skilled in the art are familiar with the basic configuration of an internet telephony system. Architectural elements and functions suitable for use in one such system are described, for instance, by the H.323 standard for multimedia transmissions, as published by the International Telecommunications Union (ITU). The entirety of the H.323 standard is hereby incorporated herein by reference. The present invention, however, is not necessarily limited to use in the H.323 configuration but may extend to other configurations or other transmission protocols now known or later developed. For example, and without limitation, another protocol that can support internet telephony is Session Initiation Protocol, or SIP.
In general, an internet telephony system facilitates telephone communication between two or more users over a packet switched, such as an IP network for example. Each user is positioned at a telephone device (hereafter “telephone”), which is generally any communications device capable of communicating real-time media signals such as speech, audio and/or video, for example. By way of example and without limitation, the telephone device may be a conventional analog telephone (e.g., a “black box telephone”), a digital telephone, a videophone, and/or a multi-media personal computer. Each telephone device (and/or telephone number) is then typically served by a network access server, which provides connectivity to the packet switched network. In the context of internet telephony, the network access server may be referred to as an internet telephony gateway (“ITG” or “gateway”) and is typically owned and operated by an internet telephony service provider (ITSP).
Alternatively, the telephone device itself may provide connectivity with the packet switched network and may serve other gateway-functions as well. Such a telephone device may be referred to as an “internet telephone” and may take any of a variety of forms now known or later developed.
To place a call over a packet switched network via an initiating gateway, a user at an initiating telephone device may establish a connection with the initiating gateway via a suitable communications link such as the public switched telephone network (PSTN) and/or other circuit switched or packet switched network or direct link. The communications link may be a permanent or semi-permanent connection (as in the case of a LAN connection between the telephone device and the gateway), which may facilitate direct dialing. Alternatively, to connect with the gateway, the user may need to place a call to the gateway via the public switched telephone network, such as by dialing a telephone number designated by the user's ITSP. In any event, the user may specify the telephone number of the called party.
The gateway serves as an interface between the packet switched network and the communications link, and in turn the telephone device. In this regard, for instance, the gateway typically performs translation between protocols, data formats and media types, to facilitate communication of information between two possibly different types of networks or links. For example, a gateway may be configured to receive a real-time media stream from the telephone device via the communications link and to encode (e.g., compress and packetize) the stream into a sequence of packets for transmission over the packet switched network to a remote destination. Similarly, a gateway may be configured to decode (e.g., de-packetize and decompress) data arriving from the packet switched network and to forward the resulting media stream via a communications link to a specified telephone device.
In a packet switched network, the location of each gateway and other element is identified by a network address. Therefore, provided with the telephone number of a called party, the initiating gateway must identify the network address of a terminating gateway that can serve the called number. To identify the network address, the gateway may query an address mapping database or may communicate with another device or process in the internet telephony system to obtain the necessary address based on the dialed number.
Given the network address of the terminating gateway, the initiating gateway may then contact the terminating gateway via the packet switched network and notify the terminating gateway of the desire to establish a connection with the called party. The terminating gateway may then establish an appropriate connection (e.g., over a communications link such as the PSTN) with a telephone device at the called number and notify the initiating gateway that the call can proceed. With the end-to-end connection thus established between the calling and called parties, the parties may then communicate with each other over the packet switched network, sending and receiving various communications signals, such as voice, video, audio and/or data.
In any telephony system, a signaling system must be provided in order to facilitate various functions involved in setting up and conducting calls. By way of example and without limitation, these functions might include monitoring the status of telephone lines to determine whether they are busy, idle or requesting service, establishing caller authorization and accounting, and sending routing and destination information throughout the system such as to alert devices in the system that a call is incoming and to establish an appropriate path of communication.
The signaling system typically provides for communication of signaling messages among elements of the telephony system. The signaling messages are used to convey signaling information such as requests, responses and status information. In addition, the signaling system typically includes a device or process to carry out signaling functions associated with various messages. For purposes of this description, this processing unit may be referred to generally as a “signaling server” or “gatekeeper.” In most cases, the signaling sever includes a database system or is configured to access one or more back-end database servers that can be used to facilitate various signaling functions.
In the context of internet telephony, for example, a variety of signaling functions are typically performed to facilitate call setup and communication. In terms of call setup, for instance, these signaling functions may include call authorization (e.g., determining whether the caller's account is current and valid, such as ensuring that the caller has paid all past bills or verifying a caller's personal identification number or security code), call accounting (e.g., notifying a billing entity that a call is being placed, in order to enable a service provider to charge for the call), address translation (e.g., identifying a network address of a terminating gateway that can serve the telephone number dialed by the caller) and establishing call connection (e.g., ensu
Grabelsky David A.
Kostas Thomas J.
McDonnell & Boehnen Hulbert & Berghoff
UTStarcom Incorporated
Yao Kwang Bin
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